This invention generally relates to access control to a network. More particularly, the present invention provides a mechanism for controlling access to a radio network based upon the current loading of the network.
A fundamental principle in the design of mobile wireless systems is that the radio spectrum is the scarcest resource. Accordingly, the network should be dimensioned in such a way that resources within the network are always available. In second-generation systems, such as Global System for Mobile Communication (GSM), which are typically optimized for speech-like services, network dimensioning to provide available resources is simple to achieve when the transport is based on STM (Synchronous Transport Mechanism) circuits. For each radio channel, a timeslot is assigned on the STM circuit to match the bit rate of the radio channel. The quality of service (QoS) can be guaranteed, but statistical multiplexing can not be used to save transport resources. This limitation on the use of statistical multiplexing is not a significant problem when the variance in bit rate is moderate, as it is in the case when speech is the dominating service.
When introducing packet switched services, where data rates vary in a greater span (for example, up to 384 kbps), a packet switched transport network is introduced to efficiently handle the variable bit rate services as well as speech. However, to dimension a packet switched transport network and still maintain the principle that the radio spectrum is the scarcest resource is not an easy task. The transmission links to the base station sites are often expensive, so over-provisioning is not necessarily the best option, especially if bandwidth can be saved by introducing some degree of resource control. Introducing QoS requirements on user connections, as opposed to best effort, makes dimensioning even harder. Admission control is needed when there are no transport resources available. After all, it is better to give a busy tone than to establish the call with a bad quality, since the user pays to get an expected quality of service.
As such, it is essential that we have a simple and scalable resource management scheme for realtime traffic in a packet switched network. In order for real-time services, such as voice, to function satisfactorily in an IP-based radio access network (RAN), for example, there need to be adequate transport resources in the RAN to handle the particular instance of that service (e.g., a phone call).
The Differentiated Services (DiffServ) working group of the Internet Engineering Task Force (IETF) has established scalable QoS mechanisms, commonly known as Differentiated Services, which have now been implemented by various router vendors. DiffServ is defined by IETF RFC 2474, and it is expected that DiffServ will be the primary mechanism for implementing QoS mechanisms in IP-based networks.
An IP network that includes DiffServ functionality is called the DS domain and consists of boundary nodes and interior nodes. The boundary nodes typically have full QoS functions, while the interior nodes have limited QoS functions. Full QoS functionality includes packet classification, during which each incoming packet is classified into a DiffServ Codepoint (DSCP) that is marked in the IP header. Full QoS functionality also includes the policing and shaping of the incoming packets, so that the bandwidth of each QoS class (or DSCP) may be kept within configured bounds.
The interior router forwards packets according to the Per-Hop Behavior (PHB) that the given DSCP value is mapped to. By using several different Per-Hop Behaviors in an interior router, QoS differentiation is provided. Examples of Per-Hop Behaviors specified by IETF are Assured Forwarding (AF) (RFC 2597) and Expedited Forwarding (EF) (RFC 2598).
As an example of a cellular radio access network, we describe the RAN for Global System for Mobile Communication (GSM). The GSM RAN includes a number of different kinds of nodes, some of which are illustrated in FIG. 1.
The BTS (Base Transceiver Station or “base station”) includes the RF (Radio Frequency) functionality and terminates the IP tunneling layer. The area covered by one BTS is defined as a cell. Several BTSs can be co-located, sharing the same antenna on the same base station site. The transport between the BTS and BSC (Base Station Controller) carries primarily airframes, which are tunneled through the IP network. These networks are large both in terms of the number of nodes as well as the geographic size. Many thousands of BTSs and BSCs could potentially be interconnected.
The transport from the BTS to the BSC is the part of the network that is most sensitive to delays and has the highest volume of real-time traffic. In some configurations, the amount of real-time traffic corresponds to the amount of voice traffic, and the network must ensure appropriate QoS for approximately 90% voice traffic.
The traffic volume for voice carried in the network can vary from a few calls up to fifty voice calls per BTS, and up to several thousand simultaneous calls (Erlang) per BSC site. In this case, several BSCs may be co-located at the same site.
The transmission between BTSs (due to the wide area coverage of the cellular network) and the BSC is often on leased lines, which may be very expensive when compared to the cost of transmission in the backbone. Even if the cost for leased lines decreases over the years, the “last mile” to the BTS is likely to continue to be expensive when the BTS is located remotely (e.g., on a mountaintop). Dimensioning using over-provisioning might therefore be prohibitively expensive. As such, mechanisms that can be used to optimize the utilization of available bandwidth in these expensive links is very important. Dynamic allocation of resources and optimization of bandwidth to reduce the cost is, therefore, an important feature.
In addition to traffic volume, mobility can significantly impact network resources. Handover (or handoff) is the process, generated by mobility, of establishing a radio link in a MC new cell and releasing the radio link in the old cell. In the GSM context, mobility usually generates handover for voice traffic an average of one to two times per call. For third generation networks, such as WCDMA and cdma2000, where it is necessary to keep radio links to several cells simultaneously to provide macrodiversity, the handover rate is typically much higher. Therefore, because of the handover rate, the admission control process has to cope with far more admission requests than call setups alone would generate.
Handover can also result in packet loss. If the processing of an admission request causes a delayed handover to the new BTS, some packets might be discarded, and the overall speech quality might be degraded significantly. Also, a delay in handover may cause degradation for other users. This is especially true for systems using macrodiversity and frequency reuse in every cell, where a handover delay will cause interference for other users in the same cell. Further, in the worst case, a delay in handover may cause the connection to be dropped, especially if the handover was made due to bad radio link quality.
Therefore, it is critical that an admission control request for handover be carried out very quickly. Since the processing of an admission control request is only one of many tasks performed during handover, the time to perform admission control should be a fraction of the time available for handover and may be on the order of 50 ms or less. This requirement will, of course, have a major influence on the architecture of resource management of the IP-based cellular access network.
The bandwidth broker performs the task of admission control for the packet switched (IP-based) transport network. It is believed that by introducing a bandwidth broker into the architecture, transmission costs can be saved by reducing the bandwidth margins while still maintaining quality of service.
Accordingly, there is a need to provide a scalable admission control process having a fast response time.